It is demanded in a mobile communication system that speech signals are compressed to low bit rates to transmit to efficiently utilize radio wave resources and so on. On the other hand, it is also demanded that quality improvement in phone call speech and call service of high fidelity be realized, and, to meet these demands, it is preferable to not only provide quality speech signals but also encode other quality signals than the speech signals, such as quality audio signals of wider bands.
The technique of integrating a plurality of coding techniques in layers is promising for these two contradictory demands. This technique combines in layers the first layer for encoding input signals in a form adequate for speech signals at low bit rates and a second layer for encoding differential signals between input signals and decoded signals of the first layer in a form adequate to other signals than speech. The technique of performing layered coding in this way have characteristics of providing scalability in bit streams acquired from an encoding apparatus, that is, acquiring decoded signals from part of information of bit streams, and, therefore, is generally referred to as “scalable coding (layered coding).”
The scalable coding scheme can flexibly support communication between networks of varying bit rates thanks to its characteristics, and, consequently, is adequate for a future network environment where various networks will be integrated by the IP protocol.
For example, Non-Patent Document 1 discloses a technique of realizing scalable coding using the technique that is standardized by MPEG-4 (Moving Picture Experts Group phase-4).
This technique uses CELP (Code Excited Linear Prediction) coding adequate to speech signals, in the first layer, and uses transform coding such as AAC (Advanced Audio Coder) and TwinVQ (Transform Domain Weighted Interleave Vector Quantization) with respect to residual signals subtracting first layer decoded signals from original signals, in the second layer.
By contrast with this, Non-Patent Document 2 discloses a method of encoding MDCT coefficients of a desired frequency bands in layers using TwinVQ that is applied to a module as a basic component. By sharing this module to use a plurality of times, it is possible to implement simple scalable coding of a high degree of flexibility. Although this method is based on the configuration where subbands which are the targets to be encoded by each layer are determined in advance, a configuration is also disclosed where the position of a subband, which is the target to be encoded by each layer, is changed within predetermined bands according to the property of input signals.    Non-Patent Document 1: “All about MPEG-4,” written and edited by Sukeichi MIKI, the first edition, Kogyo Chosakai Publishing, Inc., Sep. 30, 1998, page 126 to 127    Non-Patent Document 2: “Scalable Audio Coding Based on Hierarchical Transform Coding Modules,” Akio JIN et al., Academic Journal of The Institute of Electronics, Information and Communication Engineers, Volume J83-A, No. 3, page 241 to 252, March, 2000    Non-Patent Document 3: “AMR Wideband Speech Codec; Transcoding functions,” 3GPP TS 26.190, March 2001.    Non-Patent Document 4: “Source-Controlled-Variable-Rate Multimode Wideband Speech Codec (VMR-WB), Service options 62 and 63 for Spread Spectrum Systems,” 3GPP2 C.S0052-A, April 2005.    Non-Patent Document 5: “7/10/15 kHz band scalable speech coding schemes using the band enhancement technique by means of pitch filtering,” Journal of Acoustic Society of Japan 3-11-4, page 327 to 328, March 2004